1. In the "Record" screen in which gain and level meters are shown, would you consider adding a a display of maximal levels (i.e. a peak meter). The output screen on my old Technics tape recorder does this ... the display shows levels but maximal levels remain illuminated for a brief perod of time.
This is useful for monitoring of a recording that differs markedly in levels over short periods of time. Watching the peak meter can help one to avoid clipping.

2. I know that one can add to an existing file by copy and paste operations, but it would be great to be able to have menu (or context menus) to:

1. In the "Record" screen in which gain and level meters are shown, would you
consider adding a a display of maximal levels (i.e. a peak meter). The output
screen on my old Technics tape recorder does this ... the display shows levels
but maximal levels remain illuminated for a brief perod of time.
This is useful for monitoring of a recording that differs markedly in levels
over short periods of time. Watching the peak meter can help one to avoid
clipping.

Not to try to tell anyone here how to record, but here is a tip that should
prevent anyone from ever again clip while recording.

Digital sound being non-hissy, it responds perfectly to increases in signal
level; in other words, one can boost signal without concern of boosting
hiss, or other input noise. Thus, one can record at a relatively very low
level, in other words a level that you are virtually certain will not clip
anywhere, and boost it to a more desirable level in editing. As best I
understand it, there is no downside to working in this manner.

1. In the "Record" screen in which gain and level meters are shown, would you
consider adding a a display of maximal levels (i.e. a peak meter). The output
screen on my old Technics tape recorder does this ... the display shows levels
but maximal levels remain illuminated for a brief perod of time.
This is useful for monitoring of a recording that differs markedly in levels
over short periods of time. Watching the peak meter can help one to avoid
clipping.

I’ve been asking for this for years… ideally with options for how long the peak is held, including infinitely, and separately for each channel. This lack on the part of Amadeus is one reason i keep using Coaster 1.1.3 on Mac OS 9, as all Coaster does is record, with wonderful meters that suit my needs very well. As Coaster is no longer being developed and its author actually recommends Amadeus, i really wish Martin and the Coaster author could get together to share code, so Amadeus can rise to the next level of greatness.

pm@philxmilstein.com wrote:

smithsm wrote:
Not to try to tell anyone here how to record, but here is a tip that should
prevent anyone from ever again clip while recording.

Digital sound being non-hissy, it responds perfectly to increases in signal
level; in other words, one can boost signal without concern of boosting
hiss, or other input noise. Thus, one can record at a relatively very low
level, in other words a level that you are virtually certain will not clip
anywhere, and boost it to a more desirable level in editing. As best I
understand it, there is no downside to working in this manner.

Actually, Phil, there is a decided downside, as i understand things: quantization noise. Linear PCM recording, which is what i believe is generally used and is used in Amadeus, has lower effective resolution at lower signal levels. With “loud” signals closer to the 0 dB maximum, there are more available numeric representations for each little bit of the waveform. At the other extreme, very close to the lowest possible signal level that can be recorded at the current bit level (i am assuming 16 bit resolution here, which is what i use), there are not that many possible representations for waveform details, and there can be digital “graininess” from these limits. (Quantization noise is related to this, yet my brain is too fuzzy to explain this more accurately… maybe someone else here will do better.)

There is a lot of signal range between 0 dBFS and where quantization issues are likely to be a problem, so while my understanding is that the above is a real-world consideration, it may not be significant in practice for what most of us are doing. More important in the real world (in my experience so far) is the limit of one’s A-D converter. Something like a Griffin iMic or the internal A-D of older Macs (and maybe current ones, which i do not own) is going to limit the actual s/n ratio to about 53 dB (this is for my Power Mac 8600/300, as interpreted by the metering in Coaster), well below the stated specification (this number is unweighted and similar numbers have been verified by me on other PowerSurge series Macs, and a circa 2002 iMic).

The limits here tend to be inadequacies in the analog section of these devices. Users with higher-end A-D may have a much greater unweighted s/n and then Phil’s suggestion is more reasonable as reasonable lowering of recording levels should still be sufficiently above the quantization noise for the latter to be a non-issue (with typical material with a dynamic range in the 40 to 50 dB range, if that!).

Now, the material i am recording is typically from analog cassettes or vinyl phonograph records. Unweighted s/n for analog cassettes recorded and reproduced on a Nakamichi or similar high-end cassette deck with brand-name tape (TDK is what i used usually) where the machine(s) was/were hand-calibrated by me (professional home audio equipment repair technician between 1981 and 1995) so that the Dolby or dbx would operate properly had s/n ratios between about 42 and maybe 54 dB, with mine usually averaging 45 to 47 dB (TDK D, Dolby B). Unweighted s/n for vinyl used to be reported as about 65 dB, which i understand was for decent-grade vinyl in as-new condition.

Now, s/n, which i should really be typing as S/N, for the analog formats does not include the “over 0 dB” range where these formats are still usable, which is wholly inconsistent yet the standard assumptions have been +8 dB to +16 dB. The latter assumption is the one used by Sony/Philips for the original CD-DA audio CD format: standard analog line level for home audio systems has been -10 dBV (0.3V) for decades. The first few generations of CD players emitted +6 dBV (2V) with 0 dBFS test tones on CD (and i expect DVD players playing the same audio CD test discs still do this, though i have not checked).

For the analog formats, +4 to +8 dB should be sufficient, so we are looking at dynamic ranges of up to 73 dB (-65 dB S/N and +8 dB headroom, for clean good-quality vinyl phonograph records). With an iMic or a Mac like mine limiting this to a 53 dB dynamic range, levels are hypercritical and need to be as high as possible! The only reason i can even record vinyl on my system is that the “pop” (punk, early electronica, new wave, etc. etc. etc.) program material is nowhere close to requiring even 50 dB of dynamic range, so it will still “fit” if i choose my levels carefully. Yet, i do not have the luxury of running my maximum levels 10 or 15 or 20 dB below 0 dBFS to guarantee no clipping then amplifying later (and it may be that Phil does not recommend this low a level… he cited no numbers).

So… for my system with limited A-D range, i need very accurate metering, and as i am recording already-recorded material (nothing live), i have the luxury of getting the levels exactly correct. I set my goal as having a maximum peak level for an LP side of 0 to -0.3 dB on the loudest channel. So far, this works wonderfully. Usually the level stays there, yet if necessary i can reduce it on a copy of the file if needed for the average level to match other program material with a wider dynamic range. There Phil is 100% correct: i can attenuate the volume digitally with no meaningful loss in sound quality.

BTW, my level reference is for the desired program material, not including clicks and pops and other sonic anomalies. Doesn’t matter if those go well past 0 dBFS, and they usually do: they are severe distortion already, and they are most likely going to be removed._________________))Sonic((

Digital sound being non-hissy, it responds perfectly to increases in signal
level; in other words, one can boost signal without concern of boosting
hiss, or other input noise. Thus, one can record at a relatively very low
level, in other words a level that you are virtually certain will not clip
anywhere, and boost it to a more desirable level in editing. As best I
understand it, there is no downside to working in this manner.

What you described is the best way, but you still don't want to record
at any lower level than necessary. Yes, definitely leave headroom. But
going too low means throwing away resolution, which absolutely will
affect the sound quality.

Thanks, Sonic Purity
A great summary of the differences between analog and digital! (So about 16 dB of the higher dynamic range of the cd turn out to be a clever marketing trick.)

I’m glad if anything was useful and no one’s eyeballs fell out .

A matter of perspective in terms of “16 dB of the higher dynamic range of the cd turn out to be a clever marketing trick.” IMO, they were doing the best they could to match the new digital concept of a brick wall for headroom with the older analog concept of a fuzzy and varying ceiling for headroom. Their goal was to have people able to plug their audio CD players into their existing analog audio systems and have levels that were consistent with other signal sources of the time (FM, cassette, reel tape, turntable, etc.).

I really don’t know where they came up with the +16 dB headroom number, though in practice it seems like a valid choice. What actually happened was that recording engineers seemed to be doing what i recommend in terms of keeping their levels as high as possible, and at the same time a lot of program material was pre-compressed for optimal playback in an automobile environment. What i believe Sony and Philips had in mind with compressed material like this is that it would be digitally attenuated for an average level of -10 dBV, as was typical for other sources. Yet there was no firm standard for this, or if there was, it was ignored: some labels did this, others ran the levels as high as possible. Result: inconsistent CD to CD and CD to other source levels. Other technologies such as vinyl records and cassette tapes had physical reference levels (3.54 cm/s velocity at 1kHz? for vinyl and 200 nWb/m fluxivity [Dolby 0 level] for cassettes) for which there is no equivalent in the CD format, since the analog reference levels are for average levels, and the only CD reference is the all-bits 0 dBFS absolute peak ceiling reference.

And the situation does not seem much better 20 years later with other digital audio files, hence various auto-levelers for iTunes and so forth. Fortunately we have really useful tools like Amadeus to measure, analyze, and adjust levels!

****
Thanks to rfwilmut for the mention of ProLevel. A quick look at its home page shows great promise. Too bad the author chose not to make it freeware, as Coaster is/was for the Vintage Mac OS._________________))Sonic((